5b6cc53a2e
split from https://webrtc-review.googlesource.com/c/src/+/404061 see there for full history and manual changes Bug: webrtc:424706384 Change-Id: I113bc2dab317c3b78b79acbffd2c62fad1d85ea0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/405042 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45445}
54 lines
2.2 KiB
C++
54 lines
2.2 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/test/runtime_setting_util.h"
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#include "api/audio/audio_processing.h"
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#include "modules/audio_processing/test/protobuf_utils.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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void ReplayRuntimeSetting(AudioProcessing* apm,
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const audioproc::RuntimeSetting& setting) {
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RTC_CHECK(apm);
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// TODO(bugs.webrtc.org/9138): Add ability to handle different types
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// of settings. Currently CapturePreGain, CaptureFixedPostGain and
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// PlayoutVolumeChange are supported.
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RTC_CHECK(setting.has_capture_pre_gain() ||
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setting.has_capture_fixed_post_gain() ||
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setting.has_playout_volume_change());
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if (setting.has_capture_pre_gain()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCapturePreGain(
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setting.capture_pre_gain()));
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} else if (setting.has_capture_fixed_post_gain()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(
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setting.capture_fixed_post_gain()));
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} else if (setting.has_playout_volume_change()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(
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setting.playout_volume_change()));
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} else if (setting.has_playout_audio_device_change()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreatePlayoutAudioDeviceChange(
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{.id = setting.playout_audio_device_change().id(),
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.max_volume =
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setting.playout_audio_device_change().max_volume()}));
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} else if (setting.has_capture_output_used()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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setting.capture_output_used()));
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}
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}
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} // namespace webrtc
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